13 Oct
2011
13 Oct
'11
10:02 p.m.
Hello,
On 10/10/11 6:43 AM, Austin Einter wrote:
If the SIP UA is not behind the NAT, and Kamailio
insersts
a=nortpproxy:yes in SDP body, can it cause one sided audio.
The a= line should be
added only when force_rtp_proxy() is used, which
should happen when is a call that involves NAT, unless you call that
function for all calls.
Use ngrep on kamailio server to watch traffic on port 5060 and check if
the sdp for INVITE and 200 OK have the proper IP addresses -- they
should be rtpproxy ip address when invoking force_rtp_proxy().
Cheers,
Daniel
I am using PJSIP as my UA, kamailio 3.1.5 , rtp proxy
1.2.1 as
intermediate proxy for signalling and media. In kamailio config, I
have enabled WITH_NAT.
On call setup, only one way audio is there. When I looked at SIP
messages, the only additional stuff I can see a=nortpproxy and nat=yes
in recpord route.
Without kamailio as intermediate proxy, audio is good both way.
So I feel, when I bring up kamailio as intermediate proxy I am doing
some mistake configuration wise either at PJSIP UA or in Kamailio.
Has anybody faced this issue previously.
If so please share how can we fix this issue.
Thanks in advance
Austin
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda